U.S. patent application number 16/258015 was filed with the patent office on 2019-05-23 for method and apparatus for in-ear canal sound suppression.
This patent application is currently assigned to Staton Techiya, LLC. The applicant listed for this patent is Staton Techiya, LLC. Invention is credited to Marc Boillot, Steven Goldstein, Jason McIntosh, John Usher.
Application Number | 20190158672 16/258015 |
Document ID | / |
Family ID | 66532670 |
Filed Date | 2019-05-23 |
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United States Patent
Application |
20190158672 |
Kind Code |
A1 |
McIntosh; Jason ; et
al. |
May 23, 2019 |
Method and Apparatus for in-Ear Canal Sound Suppression
Abstract
A method and system of conferencing can include the steps of
initiating a conference call at a communication device with two or
more communication devices and selecting to suppress a voice
communication of at least one communication device on the
conference call where a modified electronic signal is generated
with the selected at least one communication device so that the
voice communication from the selected at least one communication
device is inaudible. The method or system further includes sending
the modified electronic signal to at least one other communication
device on the conference call. Other embodiments are disclosed.
Inventors: |
McIntosh; Jason; (Sugar
Hill, GA) ; Boillot; Marc; (Plantation, FL) ;
Usher; John; (Beer, GB) ; Goldstein; Steven;
(Delray Beach, FL) |
|
Applicant: |
Name |
City |
State |
Country |
Type |
Staton Techiya, LLC |
Delray Beach |
FL |
US |
|
|
Assignee: |
Staton Techiya, LLC
Delray Beach
FL
|
Family ID: |
66532670 |
Appl. No.: |
16/258015 |
Filed: |
January 25, 2019 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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14943001 |
Nov 16, 2015 |
10194032 |
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16258015 |
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12245316 |
Oct 3, 2008 |
9191740 |
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14943001 |
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12115349 |
May 5, 2008 |
8081780 |
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12245316 |
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60916271 |
May 4, 2007 |
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Current U.S.
Class: |
1/1 |
Current CPC
Class: |
H04R 3/02 20130101; H04R
1/1016 20130101; H04R 2499/13 20130101; H04R 2201/107 20130101;
H04R 1/1083 20130101; H04M 3/568 20130101; H04R 3/005 20130101;
H04M 9/08 20130101; H04R 2460/01 20130101; H04R 2430/01 20130101;
H04R 2460/05 20130101; H04R 2410/05 20130101 |
International
Class: |
H04M 3/56 20060101
H04M003/56; H04R 3/02 20060101 H04R003/02; H04R 1/10 20060101
H04R001/10; H04R 3/00 20060101 H04R003/00 |
Claims
1. A first communication device, comprising: a memory that stores
instructions; and a processor that executes the instructions to
perform operations, the operations comprising: selecting to
suppress a voice communication of a second communication device on
a call with the first communication device; producing acoustic
audio content based on audio content associated with the call that
is delivered via an ear canal receiver of the first communication
device; providing, by utilizing an ear canal microphone of the
first communication device, an electronic internal signal based on
the acoustic audio content to an adaptive filter coupled to a
reference signal; and generating, by utilizing the adaptive filter,
a modified electronic signal based on the electronic internal
signal and the reference signal; and suppressing the voice
communication of the second communication device in the modified
electronic signal.
2. The first communication device of claim 1, wherein the
operations further comprise providing the modified electronic
signal to a third communication device on the call after
suppressing the voice communication in the modified electronic
signal.
3. The first communication device of claim 1, wherein the
operations further comprise initiating the conference call using
the first communication device, the second communication device, or
a combination thereof.
4. The first communication device of claim 1, wherein the
operations further comprise enabling a conference call initiator to
preselect a caller to suppress.
5. The first communication device of claim 1, wherein the
operations further comprise adjusting volume of the audio content
to a predetermined level.
6. The first communication device of claim 1, wherein the
operations further comprise suppressing, by utilizing an echo
suppressor, feedback associated with a spoken voice received from
the ear canal receiver.
7. The first communication device of claim 1, wherein the
operations further comprise determining a background noise level of
an environment associated with the first communication device.
8. The first communication device of claim 1, wherein the
operations further comprise mixing an audio signal captured at an
ambient sound microphone of the first communication device with an
audio signal captured at the ear canal microphone of the first
communication device to produce a mixed signal.
9. The first communication device of claim 8, wherein the
operations further comprise providing the mixed signal to a voice
recognition system, a voice dictation system, a voice recorder, or
a combination thereof.
10. The first communication device of claim 8, wherein the
operations further comprise producing an echo estimate based on the
electronic internal signal and the mixed signal.
11. A method, comprising: selecting to suppress a voice
communication of a second communication device on a call with a
first communication device; generating acoustic audio content based
on audio content associated with the call that is delivered via an
ear canal receiver of the first communication device; providing, by
utilizing an ear canal microphone of the first communication
device, an electronic internal signal based on the acoustic audio
content to an adaptive filter coupled to a reference signal; and
generating, by utilizing the adaptive filter, a modified electronic
signal based on the electronic internal signal and the reference
signal; and suppressing the voice communication of the second
communication device in the modified electronic signal.
12. The method of claim 11, further comprising modeling an ear
canal transfer function by utilizing the adaptive filter.
13. The method of claim 11, further comprising providing a voice
associated with the call in the modified electronic signal.
14. The method of claim 11, further comprising managing an echo
feedback condition affecting a voice activity level determined in
an environment associated with the first communication device.
15. The method of claim 11, further comprising subtracting an echo
estimate from the electronic internal signal to produce the
modified electronic signal.
16. The method of claim 11, further comprising further comprise
suppressing feedback associated with a spoken voice received from
the ear canal receiver.
17. The method of claim 11, further comprising transmitting the
modified electronic signal to a third communication device.
18. The method of claim 11, further comprising preserving a spoken
voice occurring during the call when generating the modified
electronic signal.
19. The method of claim 11, further comprising adapting a first set
of filter coefficients of a Least Mean Squares filter to model an
inner ear-canal microphone transfer function of a first user
associated with the first communication device.
20. A first communication device, comprising: a memory that stores
instructions; and a processor that executes the instructions to
perform operations, the operations comprising: producing acoustic
audio content based on audio content associated with a call that is
delivered via an ear canal receiver of a first communication device
on a call with a second communication device; providing, by
utilizing an ear canal microphone of the first communication
device, an electronic internal signal based on the acoustic audio
content to an adaptive filter coupled to a reference signal; and
generating, by utilizing the adaptive filter, a modified electronic
signal based on the electronic internal signal and the reference
signal; and suppressing a voice communication of the second
communication device in the modified electronic signal.
Description
CROSS-REFERENCE TO RELATED APPLICATION(S)
[0001] This application is a Continuation of U.S. patent
application Ser. No. 14/943,001, filed on Nov. 16, 2015, which is a
Continuation of U.S. patent application Ser. No. 12/245,316, filed
on Oct. 3, 2008, now U.S. Pat. No. 9,191,740, which is a
Continuation-In-Part of U.S. patent application Ser. No.
12/115,349, filed on May 5, 2008, now U.S. Pat. No. 8,081,780,
which claims the priority benefit of U.S. Provisional Application
Ser. No. 60/916,271, filed on May 4, 2007, the entire disclosure of
which is incorporated herein by reference. This application is also
related to U.S. patent application Ser. No. 12/110,773, filed on
Apr. 28, 2008, now U.S. Pat. No. 8,577,062, which claims the
priority benefit of U.S. Provisional Application Ser. No.
60/914,318, filed Apr. 27, 2007, the entire disclosure of which is
incorporated herein by reference. This application is also related
to U.S. patent application Ser. No. 12/170,171, filed on Jul. 9,
2008, now U.S. Pat. No. 8,526,645, the entire disclosure of which
is incorporated herein by reference.
FIELD OF THE INVENTION
[0002] The present invention pertains to sound reproduction, sound
recording, audio communications and hearing protection using
earphone devices designed to provide variable acoustical isolation
from ambient sounds while being able to audition both environmental
and desired audio stimuli. Particularly, the present invention
describes a method and device for suppressing echo in an ear-canal
when capturing a user's voice when using an ambient sound
microphone and an ear canal microphone.
BACKGROUND OF THE INVENTION
[0003] People use headsets or earpieces primarily for voice
communications and music listening enjoyment. A headset or earpiece
generally includes a microphone and a speaker for allowing the user
to speak and listen. An ambient sound microphone mounted on the
earpiece can capture ambient sounds in the environment; sounds that
can include the user's voice. An ear canal microphone mounted
internally on the earpiece can capture voice within the ear canal;
sounds generated when the user is speaking.
[0004] An earpiece that provides sufficient occlusion can utilize
both the ambient sound microphone and the ear canal microphone to
enhance the user's voice. An ear canal receiver mounted internal to
the ear canal can loopback sound captured at the ambient sound
microphone or the ear canal microphone to allow the user to listen
to captured sound. If the earpiece is however not properly sealed
within the ear canal, the ambient sounds can leak through into the
ear canal and create an echo feedback condition with the ear canal
microphone and ear canal receiver. In such cases, the feedback loop
can generate an annoying "howling" sound that degrades the quality
of the voice communication and listening experience.
SUMMARY
[0005] In a first embodiment, a method of listening to music or
other media content during a full duplex communication event, the
method comprising the steps of delivering audio content to an ear
canal of a first user by way of an Ear Canal Receiver (ECR) to
produce an acoustic audio content where the audio content includes
music or other media content, capturing in the ear canal of the
first user by way of an Ear Canal Microphone (ECM) an electronic
signal comprising the acoustic audio content and a spoken voice of
the first user in the presence of the audio content delivered to
the ear canal, suppressing the audio content in the electronic
signal while preserving the spoken voice to produce a modified
electronic signal, and sending the modified electronic signal to at
least one other user so that the audio content is sufficiently
inaudible and the spoken voice is audible during the full duplex
communication.
[0006] In a second embodiment, a method of conferencing, the method
comprising the steps of initiating a conference call with two or
more people, selecting to suppress the voice communication of at
least one person on the conference call where a modified electronic
signal is generated with the selected at least one person voice
communication being inaudible, and sending the modified electronic
signal to at least one other person on the conference call.
[0007] In a third embodiment, a method of listening to audio
content comprising the steps of listening to audio content from a
transducer coupled to a communication device, engaging in a full
duplex voice communication with the communication device where the
voice communication and the audio content is output by the
transducer, and suppressing the audio content from a transmitted
signal from the communication device such that participants
receiving the transmitted signal hear a spoken voice of a user of
the communication device but the audio content is inaudible.
BRIEF DESCRIPTION OF THE DRAWINGS
[0008] FIG. 1 is a pictorial diagram of an earpiece in accordance
with an exemplary embodiment;
[0009] FIG. 2 is a block diagram of the earpiece in accordance with
an exemplary embodiment;
[0010] FIG. 3 is a block diagram for an acoustic management module
in accordance with an exemplary embodiment;
[0011] FIG. 4 is a schematic for the acoustic management module of
FIG. 3 illustrating a mixing of an external microphone signal with
an internal microphone signal as a function of a background noise
level and voice activity level in accordance with an exemplary
embodiment;
[0012] FIG. 5 is a more detailed schematic of the acoustic
management module of FIG. 3 illustrating a mixing of an external
microphone signal with an internal microphone signal based on a
background noise level and voice activity level in accordance with
an exemplary embodiment;
[0013] FIG. 6 is a block diagram of a system for generating
modified electronic signals in accordance with an exemplary
embodiment;
[0014] FIG. 7 is a schematic of a control unit for controlling
adaptation of a first set and second set of filter coefficients of
an echo suppressor for in-ear canal echo suppression in accordance
with an exemplary embodiment;
[0015] FIGS. 8a-8b are diagrams illustrating a conference call
initiated by a user of a communication device in accordance with an
exemplary embodiment;
[0016] FIG. 9 illustrates that a user can be listening to audio
content while in voice communication with a remote communication
device in accordance with an exemplary embodiment;
[0017] FIG. 10 is an illustration of a system for suppressing a
siren or acoustic signal from a communication signal in accordance
with an exemplary embodiment; and
[0018] FIG. 11 is a block diagram of the invisible audio system of
FIG. 10 for suppressing the siren or acoustic signal in accordance
with an exemplary embodiment.
DETAILED DESCRIPTION OF THE DRAWINGS
[0019] The following description of at least one exemplary
embodiment is merely illustrative in nature and is in no way
intended to limit the invention, its application, or uses.
[0020] Processes, techniques, apparatus, and materials as known by
one of ordinary skill in the relevant art may not be discussed in
detail but are intended to be part of the enabling description
where appropriate, for example the fabrication and use of
transducers.
[0021] In all of the examples illustrated and discussed herein, any
specific values, for example the sound pressure level change,
should be interpreted to be illustrative only and non-limiting.
Thus, other examples of the exemplary embodiments could have
different values.
[0022] Note that similar reference numerals and letters refer to
similar items in the following figures, and thus once an item is
defined in one figure, it may not be discussed for following
figures.
[0023] Note that herein when referring to correcting or preventing
an error or damage (e.g., hearing damage), a reduction of the
damage or error and/or a correction of the damage or error are
intended.
[0024] Various embodiments herein provide a method and device for
automatically mixing audio signals produced by a pair of microphone
signals that monitor a first ambient sound field and a second ear
canal sound field, to create a third new mixed signal. An Ambient
Sound Microphone (ASM) and an Ear Canal Microphone (ECM) can be
housed in an earpiece that forms a seal in the ear of a user. The
third mixed signal can be auditioned by the user with an Ear Canal
Receiver (ECR) mounted in the earpiece, which creates a sound
pressure in the occluded ear canal of the user. A voice activity
detector can determine when the user is speaking and control an
echo suppressor to suppress associated feedback in the ECR.
[0025] When the user engages in a voice communication, the echo
suppressor can suppress feedback of the spoken voice from the ECR.
The echo suppressor can contain two sets of filter coefficients; a
first set that adapts when voice is not present and becomes fixed
when voice is present, and a second set that adapts when the first
set is fixed. The voice activity detector can discriminate between
audible content, such as music, that the user is listening to, and
spoken voice generated by the user when engaged in voice
communication. The third mixed signal contains primarily the spoken
voice captured at the ASM and ECM without echo, and can be
transmitted to a remote voice communications system, such as a
mobile phone, personal media player, recording device,
walkie-talkie radio, etc. Before the ASM and ECM signals are mixed,
they can be echo suppressed and subjected to different filters and
at optional additional gains. This permits a single earpiece to
provide full-duplex voice communication with proper or improper
acoustic sealing.
[0026] The characteristic responses of the ASM and ECM filters can
differ based on characteristics of the background noise and the
voice activity level. In some exemplary embodiments, the filter
response can depend on the measured Background Noise Level (BNL). A
gain of a filtered ASM and a filtered ECM signal can also depend on
the BNL. The (BNL) can be calculated using either or both the
conditioned ASM and/or ECM signal(s). The BNL can be a slow time
weighted average of the level of the ASM and/or ECM signals, and
can be weighted using a frequency-weighting system, e.g. to give an
A-weighted SPL level (i.e. the high and low frequencies are
attenuated before the level of the microphone signals are
calculated).
[0027] At least one exemplary embodiment of the invention is
directed to an earpiece for voice operated control. Reference is
made to FIG. 1 in which an earpiece device, generally indicated as
earpiece 100, is constructed and operates in accordance with at
least one exemplary embodiment of the invention. As illustrated,
earpiece 100 depicts an electro-acoustical assembly 113 for an
in-the-ear acoustic assembly, as it would typically be placed in
the ear canal 131 of a user 135. The earpiece 100 can be an in the
ear earpiece, behind the ear earpiece, receiver in the ear,
open-fit device, or any other suitable earpiece type. The earpiece
100 can be partially or fully occluded in the ear canal, and is
suitable for use with users having healthy or abnormal auditory
functioning.
[0028] Earpiece 100 includes an Ambient Sound Microphone (ASM) 111
to capture ambient sound, an Ear Canal Receiver (ECR) 125 to
deliver audio to an ear canal 131, and an Ear Canal Microphone
(ECM) 123 to assess a sound exposure level within the ear canal
131. The earpiece 100 can partially or fully occlude the ear canal
131 to provide various degrees of acoustic isolation. The assembly
is designed to be inserted into the user's ear canal 131, and to
form an acoustic seal with the walls 129 of the ear canal at a
location 127 between the entrance 117 to the ear canal 131 and the
tympanic membrane (or ear drum) 133. Such a seal is typically
achieved by means of a soft and compliant housing of assembly 113.
Such a seal creates a closed cavity 131 of approximately 5 cc
between the in-ear assembly 113 and the tympanic membrane 133. As a
result of this seal, the ECR (speaker) 125 is able to generate a
full range frequency response when reproducing sounds for the user.
This seal also serves to significantly reduce the sound pressure
level at the user's eardrum 133 resulting from the sound field at
the entrance to the ear canal 131. This seal is also a basis for a
sound isolating performance of the electro-acoustic assembly
113.
[0029] Located adjacent to the ECR 125, is the ECM 123, which is
acoustically coupled to the (closed or partially closed) ear canal
cavity 131. One of its functions is that of measuring the sound
pressure level in the ear canal cavity 131 as a part of testing the
hearing acuity of the user as well as confirming the integrity of
the acoustic seal and the working condition of the earpiece 100. In
one arrangement, the ASM 111 can be housed in the assembly 113 to
monitor sound pressure at the entrance to the occluded or partially
occluded ear canal. All transducers shown can receive or transmit
audio signals to a processor 121 that undertakes audio signal
processing and provides a transceiver for audio via the wired or
wireless communication path 119.
[0030] The earpiece 100 can actively monitor a sound pressure level
both inside and outside an ear canal and enhance spatial and
timbral sound quality while maintaining supervision to ensure safe
sound reproduction levels. The earpiece 100 in various embodiments
can conduct listening tests, filter sounds in the environment,
monitor warning sounds in the environment, present notification
based on identified warning sounds, maintain constant audio content
to ambient sound levels, and filter sound in accordance with a
Personalized Hearing Level (PHL).
[0031] The earpiece 100 can measure ambient sounds in the
environment received at the ASM 111. Ambient sounds correspond to
sounds within the environment such as the sound of traffic noise,
street noise, conversation babble, or any other acoustic sound.
Ambient sounds can also correspond to industrial sounds present in
an industrial setting, such as factory noise, lifting vehicles,
automobiles, and robots to name a few.
[0032] The earpiece 100 can generate an Ear Canal Transfer Function
(ECTF) to model the ear canal 131 using ECR 125 and ECM 123, as
well as an Outer Ear Canal Transfer function (OETF) using ASM 111.
For instance, the ECR 125 can deliver an impulse within the ear
canal and generate the ECTF via cross correlation of the impulse
with the impulse response of the ear canal. The earpiece 100 can
also determine a sealing profile with the user's ear to compensate
for any leakage. It also includes a Sound Pressure Level Dosimeter
to estimate sound exposure and recovery times. This permits the
earpiece 100 to safely administer and monitor sound exposure to the
ear.
[0033] Referring to FIG. 2, a block diagram 200 of the earpiece 100
in accordance with an exemplary embodiment is shown. As
illustrated, the earpiece 100 can include the processor 121
operatively coupled to the ASM 111, ECR 125, and ECM 123 via one or
more Analog to Digital Converters (ADC) 202 and Digital to Analog
Converters (DAC) 203. The processor 121 can utilize computing
technologies such as a microprocessor, Application Specific
Integrated Chip (ASIC), and/or digital signal processor (DSP) with
associated storage memory 208 such as Flash, ROM, RAM, SRAM, DRAM
or other like technologies for controlling operations of the
earpiece device 100. The processor 121 can also include a clock to
record a time stamp.
[0034] As illustrated, the earpiece 100 can include an acoustic
management module 201 to mix sounds captured at the ASM 111 and ECM
123 to produce a mixed signal. The processor 121 can then provide
the mixed signal to one or more subsystems, such as a voice
recognition system, a voice dictation system, a voice recorder, or
any other voice related processor or communication device. The
acoustic management module 201 can be a hardware component
implemented by discrete or analog electronic components or a
software component. In one arrangement, the functionality of the
acoustic management module 201 can be provided by way of software,
such as program code, assembly language, or machine language.
[0035] The memory 208 can also store program instructions for
execution on the processor 121 as well as captured audio processing
data and filter coefficient data. The memory 208 can be off-chip
and external to the processor 121, and include a data buffer to
temporarily capture the ambient sound and the internal sound, and a
storage memory to save from the data buffer the recent portion of
the history in a compressed format responsive to a directive by the
processor 121. The data buffer can be a circular buffer that
temporarily stores audio sound at a current time point to a
previous time point. It should also be noted that the data buffer
can in one configuration reside on the processor 121 to provide
high speed data access. The storage memory 208 can be non-volatile
memory such as SRAM to store captured or compressed audio data.
[0036] The earpiece 100 can include an audio interface 212
operatively coupled to the processor 121 and acoustic management
module 201 to receive audio content, for example from a media
player, cell phone, or any other communication device, and deliver
the audio content to the processor 121. The processor 121
responsive to detecting spoken voice from the acoustic management
module 201 can adjust the audio content delivered to the ear canal.
For instance, the processor 121 (or acoustic management module 201)
can lower a volume of the audio content responsive to detecting a
spoken voice. The processor 121 by way of the ECM 123 can also
actively monitor the sound exposure level inside the ear canal and
adjust the audio to within a safe and subjectively optimized
listening level range based on voice operating decisions made by
the acoustic management module 201.
[0037] The earpiece 100 can further include a transceiver 204 that
can support singly or in combination any number of wireless access
technologies including without limitation Bluetooth.TM., Wireless
Fidelity (WiFi), Worldwide Interoperability for Microwave Access
(WiMAX), and/or other short or long range communication protocols.
The transceiver 204 can also provide support for dynamic
downloading over-the-air to the earpiece 100. It should be noted
also that next generation access technologies can also be applied
to the present disclosure.
[0038] The location receiver 232 can utilize common technology such
as a common GPS (Global Positioning System) receiver that can
intercept satellite signals and therefrom determine a location fix
of the earpiece 100.
[0039] The power supply 210 can utilize common power management
technologies such as replaceable batteries, supply regulation
technologies, and charging system technologies for supplying energy
to the components of the earpiece 100 and to facilitate portable
applications. A motor (not shown) can be a single supply motor
driver coupled to the power supply 210 to improve sensory input via
haptic vibration. As an example, the processor 121 can direct the
motor to vibrate responsive to an action, such as a detection of a
warning sound or an incoming voice call.
[0040] The earpiece 100 can further represent a single operational
device or a family of devices configured in a master-slave
arrangement, for example, a mobile device and an earpiece. In the
latter embodiment, the components of the earpiece 100 can be reused
in different form factors for the master and slave devices.
[0041] FIG. 3 is a block diagram of the acoustic management module
201 in accordance with an exemplary embodiment. Briefly, the
acoustic management module 201 facilitates monitoring, recording
and transmission of user-generated voice (speech) to a voice
communication system. User-generated sound is detected with the ASM
111 that monitors a sound field near the entrance to a user's ear,
and with the ECM 123 that monitors a sound field in the user's
occluded ear canal. A new mixed signal 323 is created by filtering
and mixing the ASM and ECM microphone signals. The filtering and
mixing process is automatically controlled depending on the
background noise level of the ambient sound field to enhance
intelligibility of the new mixed signal 323. For instance, when the
background noise level is high, the acoustic management module 201
automatically increases the level of the ECM 123 signal relative to
the level of the ASM 111 to create the new mixed signal 323. When
the background noise level is low, the acoustic management module
201 automatically decreases the level of the ECM 123 signal
relative to the level of the ASM 111 to create the new mixed signal
323.
[0042] As illustrated, the ASM 111 is configured to capture ambient
sound and produce an electronic ambient signal 426, the ECR 125 is
configured to pass, process, or play acoustic audio content 402
(e.g., audio content 321, mixed signal 323) to the ear canal, and
the ECM 123 is configured to capture internal sound in the ear
canal and produce an electronic internal signal 410. The acoustic
management module 201 is configured to measure a background noise
signal from the electronic ambient signal 426 or the electronic
internal signal 410, and mix the electronic ambient signal 426 with
the electronic internal signal 410 in a ratio dependent on the
background noise signal to produce the mixed signal 323. The
acoustic management module 201 filters the electronic ambient
signal 426 and the electronic internal 410 signal based on a
characteristic of the background noise signal using filter
coefficients stored in memory or filter coefficients generated
algorithmically.
[0043] In practice, the acoustic management module 201 mixes sounds
captured at the ASM 111 and the ECM 123 to produce the mixed signal
323 based on characteristics of the background noise in the
environment and a voice activity level. The characteristics can be
a background noise level, a spectral profile, or an envelope
fluctuation. The acoustic management module 201 manages echo
feedback conditions affecting the voice activity level when the ASM
111, the ECM 123, and the ECR 125 are used together in a single
earpiece for full-duplex communication, when the user is speaking
to generate spoken voice (captured by the ASM 111 and ECM 123) and
simultaneously listening to audio content (delivered by ECR
125).
[0044] In noisy ambient environments, the voice captured at the ASM
111 includes the background noise from the environment, whereas,
the internal voice created in the ear canal 131 captured by the ECM
123 has less noise artifacts, since the noise is blocked due to the
occlusion of the earpiece 100 in the ear. It should be noted that
the background noise can enter the ear canal if the earpiece 100 is
not completely sealed. In this case, when speaking, the user's
voice can leak through and cause an echo feedback condition that
the acoustic management module 201 mitigates.
[0045] FIG. 4 is a schematic 300 of the acoustic management module
201 illustrating a mixing of the electronic ambient signal 426 with
the electronic internal signal 410 as a function of a background
noise level (BNL) and a voice activity level (VAL) in accordance
with an exemplary embodiment. As illustrated, the acoustic
management module 201 includes an Automatic Gain Control (AGC) 302
to measure background noise characteristics. The acoustic
management module 201 also includes a Voice Activity Detector (VAD)
306. The VAD 306 can analyze either or both the electronic ambient
signal 426 and the electronic internal signal 410 to estimate the
VAL. As an example, the VAL can be a numeric range such as 0 to 10
indicating a degree of voicing. For instance, a voiced signal can
be predominately periodic due to the periodic vibrations of the
vocal cords. A highly voiced signal (e.g., vowel) can be associated
with a high level, and a non-voiced signal (e.g., fricative,
plosive, consonant) can be associated with a lower level.
[0046] The acoustic management module 201 includes a first gain
(G1) 304 applied to the AGC processed electronic ambient signal
426. A second gain (G2) 308 is applied to the VAD processed
electronic internal signal 410. The acoustic management module 201
applies the first gain (G1) 304 and the second gain (G2) 308 as a
function of the background noise level and the voice activity level
to produce the mixed signal 323, where
G1=f(BNL)+f(VAL) and G2=f(BNL)+f(VAL)
[0047] As illustrated, the mixed signal is the sum 310 of the G1
scaled electronic ambient signal and the G2 scaled electronic
internal signal. The mixed signal 323 can then be transmitted to a
second communication device (e.g. second cell phone, voice
recorder, etc.) to receive the enhanced voice signal. The acoustic
management module 201 can also play the mixed signal 323 back to
the ECR for loopback listening. The loopback allows the user to
hear himself or herself when speaking, as though the earpiece 100
and associated occlusion effect were absent. The loopback can also
be mixed with the audio content 321 based on the background noise
level, the VAL, and audio content level. The acoustic management
module 201 can also account for an acoustic attenuation level of
the earpiece, and account for the audio content level reproduced by
the ECR when measuring background noise characteristics. Echo
conditions created as a result of the loopback can be mitigated to
ensure that the voice activity level is accurate.
[0048] FIG. 5 is a more detailed schematic of the acoustic
management module 201 illustrating a mixing of an external
microphone signal with an internal microphone signal based on a
background noise level and voice activity level in accordance with
an exemplary embodiment. In particular, the gain blocks for G1 and
G2 of FIG. 4 are a function of the BNL and the VAL and are shown in
greater detail. As illustrated, the AGC produces a BNL that can be
used to set a first gain 322 for the processed electronic ambient
signal 311 and a second gain 324 for the processed electronic
internal signal 312. For instance, when the BNL is low (<70
dBA), gain 322 is set higher relative to gain 324 so as to amplify
the electronic ambient signal 311 in greater proportion than the
electronic internal signal 312. When the BNL is high (>85 dBA),
gain 322 is set lower relative to gain 324 so as to attenuate the
electronic ambient signal 311 in greater proportion than the
electronic internal signal 312. The mixing can be performed in
accordance with the relation:
Mixed signal=(1-.beta.)*electronic ambient
signal+(.beta.)*electronic internal signal where=(1-.beta.) is an
external gain, (.beta.) is an internal gain, and the mixing is
performed with 0<.beta.<1.
[0049] As illustrated, the VAD produces a VAL that can be used to
set a third gain 326 for the processed electronic ambient signal
311 and a fourth gain 328 for the processed electronic internal
signal 312. For instance, when the VAL is low (e.g., 0-3), gain 326
and gain 328 are set low so as to attenuate the electronic ambient
signal 311 and the electronic internal signal 312 when spoken voice
is not detected. When the VAL is high (e.g., 7-10), gain 326 and
gain 328 are set high so as to amplify the electronic ambient
signal 311 and the electronic internal signal 312 when spoken voice
is detected.
[0050] The gain scaled processed electronic ambient signal 311 and
the gain scaled processed electronic internal signal 312 are then
summed at adder 320 to produce the mixed signal 323. The mixed
signal 323, as indicated previously, can be transmitted to another
communication device, or as loopback to allow the user to hear his
or her self.
[0051] FIG. 6 is an exemplary schematic of an operational unit 600
of the acoustic management module for generating modified
electronic signals in accordance with an embodiment. The
operational unit 600 may contain more or less than the number of
components shown in the schematic. The operational unit 600 can
include an adaptive filter 610, an adaptive filter 612, and a voice
decision logic 620.
[0052] Adaptive filters 610 and 612 can be a Least Mean Squares
(LMS) or Normalized Least Mean Squares (NLMS) adaptive filter that
models an ear canal transfer function (ECTF) between the ECR 125
and the ECM 123. The adaptive filter 610 generates the modified
electronic signal, e(n) 412, which is provided as an input to the
voice decision logic 620; e(n) is also termed the error signal e(n)
of the adaptive filter 610. In an echo cancellation mode, the error
signal e(n) 412 is used to update the filter H(w) to model the ECTF
of an echo path. The error signal e(n) 412 closely approximates the
user's spoken voice signal u(n) 607 when the echo suppressor 610
accurately models the ECTF.
[0053] Alternately, a first reference signal can be provided to
adaptive filter 610 and a second reference signal can be provided
to adaptive filter 612. For example, a reference signal can be
music, media content, or a voice communication signal. The error
signal e(n) 412 for adaptive filter 610 or e(n) 618 for adaptive
filter 612 is used to update their respective filters H(w) to model
the ECTF in the signal path. The error signal e(n) 412 for adaptive
filter 610 or e(n) 618 for adaptive filter 612 will closely
approximate the corresponding acoustic reference signal as output
by ECR 125 and received ECM 123 when adaptive filters 610 and 612
accurately model the ECTF.
[0054] In the configuration shown the adaptive filter 610 (and
similarly adaptive filter 612) minimizes the error between the
filtered signal, {tilde over (y)}(n), and the electronic internal
signal, z(n), in an effort to obtain a transfer function H' which
is a best approximation to the H(w) (i.e., ECTF). H(w) represents
the transfer function of the ear canal and models the echo
response. (z(n)=u(n)+y(n)+v(n), where u(n) is the spoken voice or
the reference signal that is an internal sound 607, y(n) is the
acoustic signal output by ECR 125, and v(n) is background noise (if
present, for instance due to improper sealing)).
[0055] In the echo cancellation mode, the adaptive filter 610
monitors the mixed signal 323 delivered to the ECR 125 and produces
an echo estimate {tilde over (y)}(n) of an echo y(n) 609 based on
the captured electronic internal signal 410 and the mixed signal
323. The adaptive filter 610, upon learning the ECTF by an adaptive
process, can then suppress the echo y(n) 609 of the acoustic audio
content 603 (e.g., output mixed signal 323) in the electronic
internal signal z(n) 410. It subtracts the echo estimate {tilde
over (y)}(n) from the electronic internal signal 410 to produce the
modified electronic internal signal e(n) 412.
[0056] It should be noted that more than two adaptive filters could
be used to generate multiple modified electronic signals for using
with the earpiece or other devices coupled or paired to the
earpiece. The circuitry and process disclosed herein is not limited
to an earpiece and can be practiced in other communication devices
such as a cell phone, smart phone, PDA, laptop computer, radio
communication systems, and conferencing systems to name but a few.
In a reference suppression mode, a signal or multiple signals are
suppressed from at least one device.
[0057] Referring to FIG. 8a, a conference call is initiated by the
user of a communication device 806 in accordance with an exemplary
embodiment. Communication device 806 is coupled via a network 808
to communication devices 802 and 804. In at least one exemplary
embodiment, communication device 806 opens a channel 1 to
communicate with device 802 and a channel 2 to communicate with
device 804. Thus, device 806 receives voice communication from both
devices 802 and 804.
[0058] The reference suppression mode allows the initiator of the
conference call to selectively suppress the voice communication of
one or more participants and to selectively determine which of the
participants each person gets to hear. As shown, two participants
are engaged in the call. In at least one exemplary embodiment, the
conference call initiator pre-selects which caller is suppressed.
Referring to FIG. 8b, the conference call initiator has selected
that the user of device 804 will not hear the voice of the user of
device 802. Thus, the voice communication signal from device 802 is
the reference signal for suppression. For example, the users of
devices 804 and 806 are in discussion of a business deal. The
conference call initiator wants to better understand the
implications of the business terms as the dialogue progresses. The
user of device 802 is a business associate of the conference call
initiator and provides input or comment that can be relevant to the
negotiation. The voice communication of the user of device 802 is
suppressed by communication device 806 such that only the initiator
of the conference call is heard by the user of device 804. The
suppressed communication signal is sent to the user of device 804
through channel 2. Conversely, the communication signal sent on
channel 1 is not modified allowing the business associate to hear
the full conversation while commenting to the conference call
initiator.
[0059] Referring to FIG. 9, a user can be listening to audio
content while in voice communication with a user of device 906 in
accordance with an exemplary embodiment. The user of communication
device 906 does not hear the audio content in the voice
communication. An earpiece 902 is coupled for receiving audio
content from a media player 910. As shown, earpiece 902 receives
the media content directly from media player 910 via a wired or
wireless connection. Alternately, earpiece 902 can receive the
audio content through communication device 904. The audio content
can be music, media content, or other audio content that the user
of earpiece 902 wants to hear.
[0060] Communication device 904 is paired with earpiece 902. In at
least one exemplary embodiment, the audio content provided by media
player 910 is uninterrupted by a call from the user of
communication device 906. In at least one exemplary embodiment, the
volume of the audio content can be automatically adjusted to a
predetermined level (defined by the user) that would allow voice
communication with the user of device 906 via network 908. For
example, the volume could be lowered for the user to hear that a
call is incoming and further adjust the volume if the call is
taken. In the reference suppression mode 912, the audio content is
the reference signal. Earpiece 902 suppresses the audio content
from a communication signal being sent to the user of device 906.
Alternately, device 904 can suppress the audio content from the
transmitted communication signal if it has access to the reference
signal using the method described hereinbelow. Thus, the user of
device 904 can continue to listen to a ball game, hear a news
bulletin, or enjoy music while in communication with others and the
audio content is not transmitted to those in communication with
device 904.
[0061] Referring back to FIG. 6, a switch 624 includes an input 614
coupled to a source 1, an input 616 coupled to a source 2, an
output 626, an output 628, an output 630 and control signals 632.
Source 1 and source 2 are signals such as music, media content,
voice communication, or other audio content. Control signals
configure switch 624 for providing source 1, source 2, or a
combination of source 1 and source 2 at each output 626, 628, and
630.
[0062] In the first example disclosed above, a user of the system
initiates a conference call with two other people. Source 1 is a
voice communication from a first caller. Source 2 is a voice
communication from a second caller. The conference call initiator
sets up the call on the system so that the first caller does not
hear the voice of the second caller. The system configures switch
624 such that the first and second caller voice communication is
mixed with an electronic ambient signal 426 and is provided as
mixed signal 323 to ECR 125. ECR 125 provides the acoustic audio
content 603 that includes the voices of the first and second
callers. Thus, the call initiator hears both the callers. Switch
624 is configured such that output 628 is coupled to source 2 which
is the second caller voice communication. The second caller voice
communication is the reference signal for adaptive filter 610. The
adaptive filter 610, upon learning the ECTF by an adaptive process
can suppress the second caller voice of the acoustic audio content
603 (e.g., output mixed signal 323) in the electronic internal
signal 410 (z(n)). It subtracts the second caller estimate {tilde
over (y)}(n) from the electronic internal signal 410 to produce the
modified electronic internal signal e(n) 412. Thus, the modified
electronic signal 1 will include the voice of the initiator of the
conference call but not the second caller. The modified electronic
signal 1 is transmitted to the first caller.
[0063] The system is set up for the second caller to hear the
entire conversation. Output 630 of switch 624 couples to the
adaptive filter 612. No reference signal is provided to adaptive
filter 612 so nothing in electronic internal signal 410 is
suppressed. Alternately, adaptive filter 612 can be disabled.
Modified electronic signal 2 is electronic internal signal 410 that
includes all of the voice communication. The modified electronic
signal 2 is transmitted to the second caller.
[0064] In the second example disclosed above, a user is listening
to audio content such as music when receiving a call. The user can
listen to the audio content while taking the call without the
caller hearing the audio content. A second modified electronic
signal 2 is not required so adaptive filter 612 is disabled. Source
1 couples to a media player or other audio source. For illustrative
purposes, source 1 is music from a media player. Source 2 is the
voice communication signal from a remote caller. Control signals
632 enable switch 624 for providing source 1 and source 2 at the
output 626. The music signal and the voice communication signal at
output 626 are mixed with the electronic ambient signal 426 forming
mixed signal 323. Mixed signal 323 is provided to ECR 125. ECR 125
outputs acoustic audio content 603 which includes the music and the
voice communication from the remote caller.
[0065] Switch 624 is configured such that output 628 is coupled to
source 1 which is the music signal. The music signal is the
reference signal for adaptive filter 610. The adaptive filter 610,
upon learning the ECTF by an adaptive process can suppress the
music signal of the acoustic audio content 603 (e.g., output mixed
signal 323) in the electronic internal signal 410 (z(n)). It
subtracts the music signal estimate {tilde over (y)}(n) from the
electronic internal signal 410 to produce the modified electronic
internal signal e(n) 412. Thus, the modified electronic signal 1
will include the voice of the initiator of the conference call but
not the music. The modified electronic signal 1 is transmitted to
the first caller.
[0066] The voice decision logic 620 analyzes the modified
electronic signal 412 e(n) and the electronic ambient signal 426 to
produce a voice activity level 622, .alpha. The voice activity
level .alpha. identifies a probability that the user is speaking,
for example, when the user is using the earpiece for two way voice
communication. The voice activity level 622 can also indicate a
degree of voicing (e.g., periodicity, amplitude), When the user is
speaking, voice is captured externally from acoustic ambient signal
424 by the ASM 111 in the ambient environment and also by the ECM
123 in the ear canal. The voice decision logic 620 provides the
voice activity level .alpha. to the acoustic management module 201
as an input parameter for mixing the ASM 111 and ECM 123 signals.
Briefly referring back to FIG. 4, the acoustic management module
201 performs the mixing as a function of the voice activity level
.alpha. and the background noise level (see G=f(BNL)+f(VAL)).
[0067] For instance, at low background noise levels and low voice
activity levels, the acoustic management module 201 amplifies the
electronic ambient signal 426 from the ASM 111 relative to the
electronic internal signal 410 from the ECM 123 in producing the
mixed signal 323. At medium background noise levels and medium
voice activity levels, the acoustic management module 201
attenuates low frequencies in the electronic ambient signal 426 and
attenuates high frequencies in the electronic internal signal 410.
At high background noise levels and high voice activity levels, the
acoustic management module 201 amplifies the electronic internal
signal 410 from the ECM 123 relative to the electronic ambient
signal 426 from the ASM 111 in producing the mixed signal. The
acoustic management module 201 can additionally apply frequency
specific filters based on the characteristics of the background
noise.
[0068] FIG. 7 is a schematic of a control unit 700 for controlling
adaptation of a first set (736) and a second set (738) of filter
coefficients of the adaptive filter 610 for in-ear canal echo
suppression in accordance with an exemplary embodiment. The
description hereinbelow also applies to adaptive filter 612.
Briefly, the control unit 700 illustrates a freezing (fixing) of
weights in upon detection of spoken voice. The adaptive filter 610
resumes weight adaptation when e(n) is low, and freezes weights
when e(n) is high signifying presence of spoken voice.
[0069] When the user is not speaking, the ECR 125 can pass through
ambient sound captured at the ASM 111, thereby allowing the user to
hear environmental ambient sounds. In an echo suppression mode, the
adaptive filter 610 models an ECTF and suppresses an echo of the
mixed signal 323 that is looped back to the ECR 125 by way of the
ASM 111 (see dotted line Loop Back path). When the user is not
speaking, the suppressor continually adapts to model the ECTF. When
the ECTF is properly modeled, the adaptive filter 610 produces a
modified internal electronic signal e(n) that is low in amplitude
level (i.e, low in error). The suppressor adapts the weights to
keep the error signal low. When the user speaks, the suppressor
however initially produces a high-level e(n) (e.g., the error
signal increases). This happens since the speaker's voice is
uncorrelated with the audio signal played out the ECR 125, which
disrupts the ECTF modeling ability of adaptive filter 610.
[0070] The control unit 700 upon detecting a rise in e(n), freezes
the weights of the adaptive filter 610 to produce a fixed filter
H'(w) fixed 738. Upon detecting the rise in e(n) the control unit
adjusts the gain 734 for the ASM signal and the gain 732 for the
mixed signal 323 that is looped back to the ECR 125. The mixed
signal 323 fed back to the ECR 125 permits the user to hear
themself speak. Although the weights are frozen when the user is
speaking, a second filter H'(w) 736 continually adapts the weights
for generating a second e(n) that is used to determine presence of
spoken voice. That is, the control unit 700 monitors the second
error signal e(n) produced by the second filter 736 for monitoring
a presence of the spoken voice.
[0071] The first error signal e(n) (in a parallel path) generated
by the first filter 738 is used as the mixed signal 323. The first
error signal contains primarily the spoken voice since the ECTF
model has been fixed due to the weights. That is, the second
(adaptive) filter is used to monitor a presence of spoken voice,
and the first (fixed) filter is used to generate the mixed signal
323.
[0072] Upon detecting a fall of e(n), the control unit restores the
gains 734 and 732 and unfreezes the weights of the suppressor, and
the first filter H'(w) 738 returns to being an adaptive filter. The
second filter H'(w) 736 remains on stand-by until spoken voice is
detected, and at which point, the first filter H'(w) 738 goes
fixed, and the second filter H'(w) 736 begins adaptation for
producing the e(n) signal that is monitored for voice activity.
Notably, the control unit 700 monitors e(n) from the first filter
738 or the second filter 736 for changes in amplitude to determine
when spoken voice is detected based on the state of voice
activity.
[0073] FIG. 10 is an illustration of a system for suppressing a
siren or acoustic signal from a communication signal in accordance
with an exemplary embodiment. In general, an emergency vehicle such
as a police car, military vehicle, fire truck, or ambulance is a
noisy environment. Typically, the vehicle has a loud siren or horn
that is played periodically in an emergency situation to alert
people that the vehicle is approaching. The siren is of sufficient
loudness that it penetrates the cabin of the vehicle where it is
picked up by a communication device being used. The siren can make
the intelligibility of the speech difficult to understand on the
receiving end of the communication. For example, an ambulance
medical technician is rapidly telling a hospital the condition of a
patient while the siren is blaring in the background. Removing the
siren from the communication would allow the speech to be heard
with clarity and allow concise recognition of the information being
provided.
[0074] In at least one exemplary embodiment, a system for a vehicle
1002 comprises a transducer 1006 for emitting an acoustic signal
(e.g. siren), a cabin loudspeaker 1004, an "invisible audio" (IA)
system 1010, a communication system 1012, and a microphone 1008.
The siren acoustic component detected by cabin microphone 1008 is
removed by the "Invisible Audio" (IA) system 1010 before the voice
communication signal is transmitted to a remote party via
communication system 1012. Removing the siren or acoustic signal
increases the intelligibility of the transmitted voice
communication signal to a remote party, e.g. an emergency call
centre. In another embodiment, warning signals that can be
reproduced with cabin loudspeaker 1004, such as voice audio signals
from other calling parties or vehicle warning systems, can also be
removed (or attenuated) by the IA system 1010.
[0075] FIG. 11 is a block diagram of the invisible audio system of
FIG. 10 for suppressing the siren or acoustic signal in accordance
with an exemplary embodiment. An emergency siren audio signal 1112
can be generated either algorithmically using a digital signal
processing system, or can be acquired from computer readable
memory, and is emitted using acoustic siren transducer 1116 (e.g. a
loudspeaker mounted on the outside of the emergency vehicle). The
emergency siren audio signal 1112 is further filtered using filter
1118 and subtracted from the cabin microphone signal 1122 using
signal subtraction unit 1120, to generate a modified microphone
signal 1114. In at least one exemplary embodiment, filter 1118 is a
fixed FIR-type digital filter, where the impulse response
approximates the acoustic impulse response between the siren
transducer 1116 and cabin microphone 1122. In another exemplary
embodiment, filter 1118 is an adaptive filter, where the filter
coefficients are updated over time to approximate the acoustic
impulse response between the siren transducer 1116 and cabin
microphone 1122. The adaptive filter can be updated according to
one of many well-known adaptive filter algorithms, such as the
normalized least-mean-square algorithm (NLMS) or recursive
least-mean-square algorithm. The modified microphone signal 1114
can then be further transmitted to a remote party, such as an
emergency caller control centre with better intelligibility and
clarity.
[0076] Where applicable, the present embodiments of the invention
can be realized in hardware, software or a combination of hardware
and software. Any kind of computer system or other apparatus
adapted for carrying out the methods described herein are suitable.
A typical combination of hardware and software can be a mobile
communications device with a computer program that, when being
loaded and executed, can control the mobile communications device
such that it carries out the methods described herein. Portions of
the present method and system may also be embedded in a computer
program product, which comprises all the features enabling the
implementation of the methods described herein and which when
loaded in a computer system, is able to carry out these
methods.
[0077] While the present invention has been described with
reference to exemplary embodiments, it is to be understood that the
invention is not limited to the disclosed exemplary embodiments.
The scope of the following claims is to be accorded the broadest
interpretation so as to encompass all modifications, equivalent
structures and functions of the relevant exemplary embodiments.
Thus, the description of the invention is merely exemplary in
nature and, thus, variations that do not depart from the gist of
the invention are intended to be within the scope of the exemplary
embodiments of the present invention. Such variations are not to be
regarded as a departure from the spirit and scope of the present
invention.
* * * * *